Thanks! But where can I put the login and password of my sip account of
the gateway?
Best Regards,
Marc LEURENT
Julien REVERET a écrit :
You can redirect calls to a PSTN gateway using this kind of routing :
if (method=="INVITE")
{
if (uri=~"sip:[EMAIL PROTECTED]") # Here we check the number dialed
{
#authorize if a call is going to PSTN
if(!proxy_authorize("domain.net", "subscriber"))
{
proxy_challenge("domain.net", "0");
return;
};
xlog("L_INFO", "CALL: Call to international number\n");
rewritehostport("voip_gw.domain.net:5060"); # rewriting SIP headers
route(1);
}
By checking the uri and rewriting destination host you can route your PSTN
calls to a PSTN gateway. The gateway can be an Asterisk PBX, a SIP/PSTN
appliance or any kind of SIP provider, I guees you already know that. The
example above is taken from openser and asterisk realtime integration.
----- Original Message -----
From: "Marc LEURENT" <[EMAIL PROTECTED]>
To: [email protected]
Sent: mercredi 18 juillet 2007 10 h 02 (GMT+0100) Europe/Berlin
Subject: [OpenSER-Users] Redirect to Trunk IP SIP/PSTN gateway + billing
Does anyone succeed in redirecting SIP calls like [EMAIL PROTECTED] to
a SIP/PSTN gateway provider without using asterisk?
Thanks
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